The invention relates generally to an adaptive processor for transmitting an input signal, for instance, an telephonic audio signal, in digital form at greatly reduced bandwidth.
To transmit an input signal composed of a series of digital samples, it is not necessary to transmit the stream of digital samples itself. For example, in a simple transmitting processor, digital values are transmitted which represent only the changes in the input signal. That is, if the input signal has a constant level over some adjacent sample, digital values representing only zero, or no digital values at all, are transmitted. Although this simple technique does achieve some significant measure of channel bandwidth rejection, nevertheless, schemes which achieve a much greater reduction in channel bandwidth are known.
In one well known technique, the input signal is compressed, such as in accordance with an A-law or a .mu.-law compression characteristic curve prior to transmission. With these methods, the "delta" of the signal transmitted represents a smaller change in the actual signal at low signal levels than at high signal levels. Although these methods do achieve a significant increase in dynamic range, nevertheless, they are disadvantageous in that distortion in the signal is increased for high signal levels. Also, the channel bandwidth required is greater than what can be accepted in many instances.
Recently adaptive delta modulators have been developed. In these modulators, the quantizer step size is adjusted in dependence upon the level of the incoming signal, similar to the A-law and .mu.-law compression techniques. Some specific examples of such delta modulators will now be discussed.
In an article "A Strategy for Delta Modulation and Speech Reconstruction", J. C. Su et al., Comsat Technical Review, Vol. 6, No. 2, Fall 1976, there is described an adaptive delta modulator utilizing an adaptive slope strategy. The adaptive slope control is incorporated into a linear delta modulation system operating at 32 kbps. The linear delta modulation system consists of a bi-state comparator in a forward loop and a two-pole, one-zero linear low-pass filter in a feedback loop. The adaptive loop slope control utilizes a variable step size which, at each sampling instant, determines the step size based on the four most recent bits obtained by sampling the comparator.
A further example is disclosed in the article "Adaptive Quantization in Differential PCM Coding of Speech", P. Cummiskey, The Bell System Technical Journal, Vol. 52, No. 7, September 1973. Therein is described an adaptive differential PCM coder which makes instantaneous exponential changes of a quantizer step size. The quantizer utilizes a simple first-order predictor and a time-invariant, minimally complex adaptation strategy. Step-size multipliers depend only on the most recent quantizer output, and input signals of unknown variants can be accomodated.
A residual encoder scheme is described in the article "The Residual Encoder--An Improved ADPCM System for Speech Digitization", D. L. Crone et al., IEEE Transactions on Communications, Vol. COM-23, No. 9, September 1975. That encoder utilizes an adaptive differential pulse-code modulation (ADPCM) system including an adaptive predictor, an adaptive quantizer, and a variable length source coding scheme. A predictor value is subtracted from a sample of an input signal and the difference supplied to a quantizer. The output of the quantizer is coded and transmitted and also passed through an inverse quantizer and predictor circuit in a feedback loop to generate the predictor coefficients.
A further example of an adaptive delta modulator is described in "Adaptive Delta Modulation with a One-Bit Memory", N. S. Jayant, The Bell System Technical Journal, Vol. 49, No. 3, March 1970. That article describes a delta modulator which, at every sampling instant, adapts a step size on the basis of a comparison between the two latest channel symbols. Specifically, the ratio of a modified step size to the previous step size depends of whether the two latest channel symbols are equal or not.
U.S. Pat. No. 4,071,825 to McGuffin teaches an adaptive delta modulation system in which the input analog signal is periodically compared with an analog feedback signal to generate a digital output signal. The analog feedback signal is generated from the digital output signal by generating a signal having an amplitude indicative of the absolute value of the derivative of the analog input signal, multiplying the absolute value signal by the digital output signal, and integrating the product signal.
U.S. Pat. No. 4,123,709 to Dodds et al. relates to an adaptive digital delta modulation technique in which an input analog signal is periodically sampled and a binary bit is generated for each period with the logic level of the binary bit being dependent on whether the sampled signal is greater or smaller than an approximation of the previous sample. A decoding apparatus converts the stream of binary bits to approximate the analog signal by periodically charging or discharging a capacitor integrator by predetermined variable steps. The charging or discharging of the integrator capacitor during each period is determined by the logic level of the binary bits, whereas the increase or decrease in step size for successive periods is determined by successive similar signal binary bits and successive dissimilar bits, respectively.
U.S. Pat. No. 4,208,740 to Yin et al. discloses an adaptive delta modulation system in which a compandor executes a companding algorithm having a nonlinear characteristic to provide a rapid increase or decrease in step size to follow a fast rising start or fast decaying end of talk spurt in an input signal so as to minimize slope overload distortions and to provide a small increase or decrease in step size during the smooth portion of the talk spurts in input signals to reduce any granular noise effect. The step size generated by the compander is directly related to the input signal level, providing a maximum companding range in which signal-to-noise characteristics are preserved for the soft or low voiced speaker as well as for the loud voiced speaker and providing a fast transient response suitable for most telephone applications.
Although not directly related to the field of adaptive delta modulators, the articles "Analysis of An Adaptive Impulse Response Echo Cancellor", S. J. Campanella, Comsat Technical Review, Vol. 2, No. 1, Spring 1972, pages 1-38 and "A Twelve-Channel Digital Echo Cancellor", D. L. Duttweiler, IEEE Transactions on Communications, Vol. COM-26, No. 5, 1978, pages 647-653, are of interest in disclosing techniques for computing estimates of samples of a digitized input signal. FIG. 1 of the aforementioned Campanella et al. article shows a block diagram of a digital echo cancellor in which estimates of the echo signal are computed using a convolution process. A storage register X stores a predetermined number of receive-side signal samples, while a second storage register H stores model echo path impulse responses. The values stored in the two registers are convolved to produce the estimates of the echo signal. The Duttweiler article also discloses a digital echo cancellor utilizing a convolution technique.
Although the above-described examples of adaptive delta modulators (or processors) have achieved some significant reductions in channel bandwidth while providing generally quite good received signal quality, nevertheless, due to the ever-increasing congestion of available communications channels, it is desired to provide an adaptive processor in which the channel bandwidth can be yet further reduced without greatly affecting the perceived quality of the received signal.